GIP2000B ideally integrates with the PSTN phone and the VoIP phone together.GIP2000B complies with SIP v2.0. It supports both PSTN mode and IP mode and is equipped with three Ethernet ports for PC laptops and one WAN port for Internet connection.You can switch from one mode to another very easily.

Specification:

VoIP SoC CPU: MICREL KS8695X@133MHz
                      DSP: TI TMS3205402@100MHz

Flash 2MBytes Flash

RAM 8MBytes SDRAM    128Kbytes SRAM

Display 2 * 12 Characters LCD

Phone Faceplate 
 12 Dialing Buttons (0~9, *, #),
                      12 Fixed Function Buttons,
                      3 Memory Buttons.

Power Supply 
                      AC/DC Adapter   AC Input: 100~240V, 50~60Hz
                      DC Output: 5V, 1.2A

I/O Interfaces 
                      3 RJ-45 for 10/100 Switched Ethernet Ports,
                      One for WAN Port, The Others for LAN Port
                      1 RJ-11 for Lifeline Port
                      1 RJ-22 for Handset
                      Embedded MIC/SPEAKER

Audio Codec
 G.711 (A-Law, ?Law)
                     G.723.1
                     G.729A/AB
                     Adaptive Jitter Buffer Management
                     Voice Activity Detection
                     Comfort Noise Generation
                     Echo Cancellation

Protocols 
                    SIP V2 (RFC 3261,3262,3263,3264)
                    Backward Compatible with RFC2543
                    Session Timer (RFC4028)
                    SDP (RFC2327)
                    RTP/RTCP (RFC1889 and RFC1890)
                    NAPTR for SIP URI Lookup (RFC2915)
                    STUN (RFC 3489)
                    ARP/RARP (RFC 826/903)
                    SNTP (RFC 2030)
                    DHCP/PPPoE
                    HTTP Server for Web Management
                    TFTP/HTTP for Auto Provisioning
                    Message Waiting Indicator (RFC3842)
                    DHCP Option Codes for SIP (RFC3361)
                    DNS/DNS SRV (RFC1706 and RFC 2782)

Applications
 Mac address cloning
                     SIP proxy redundancy
                                  -dynamic via DNS SRV, A records
                     Direct IP to IP calling
                     NAT Traversal
                                   -Static NAT Route
                                   - Traversal by STUN
                     100 Entries Address Book
                     Support lifeline function

Management
 Firmware Upgradeable
                     Web Management Interface
                     IVR-driven Management Interface
                     Auto Provisioning

Call Features
                     3-way Conference
                     Music on hold
                    Call History
                          - Most Recently Missed Calls
                          - Most Recently Received Calls
                          - Most Recently Dialed Numbers
                     Call Hold
                     Call Forwarding
                     Call Mute
                     Call Park
                     Call Progress Tone
                     Call Transfer
                     Call Waiting
                     Caller ID
                     DTMF Relay
                           - In-band, Out-band(RFC2833) and SIP Info
                     Full-duplex Speakerphone
                     Hotline
                     Redial
                     Speed Dial
                     Real time fax over IP via T.38
                     Volume Adjustment
                             -Handset/Headset, Speaker and Ringer

Dimension
          (L) 180 x (W) 170 x (H) 60 mm

Weight
          605g